Voice transmission over packet networks is subject to delay variation, commonly known as jitter. Jitter may, for example, be measured in terms of inter-arrival time (IAT) variation or packet delay variation (PDV). IAT variation may be measured according to the receive time difference of adjacent packets. PDV may, for example, be measured by reference to time intervals from a datum or “anchor” packet receive time. In Internet Protocol (IP)-based networks, a fixed delay can be attributed to algorithmic, processing and propagation delays due to material and distance, whereas a variable delay may be caused by the fluctuation of IP network traffic, different transmission paths over the Internet, etc.
VoIP (voice over Internet Protocol) receivers generally rely on a “jitter buffer” to counter the negative impact of jitter. By introducing an additional delay between the time a packet of audio data is received and the time that the packet is reproduced, a jitter buffer aims at transforming the uneven flow of arriving packets into a regular flow of packets, such that delay variations will not cause perceptual sound quality degradation to the end users. Voice communication is highly delay sensitive. According to ITU Recommendation G.114, for example, one-way delay should be kept below 150 ms for normal conversation, with above 400 ms being considered unacceptable. Therefore, the additional delay added by a jitter buffer needs to be small enough to avoid causing perceptual sound quality degradation. Unfortunately, a small jitter buffer will lead to more frequent packet loss when packets arrive later than expected due to network delays.